PJSIP TLS re-invites after 15 minutes which drops the call

I am running Asterisk certified/13.13-cert3 with the embedded pjproject. When I make a TLS outbound call it sends out the orignal invite through our sip gateway – just like I would expect:
[May 9 11:07:17] INVITE sip:TLS1000000@xxxxxx.xxx.xxxxxxxxx.net:5061 SIP/2.0
After about 15 minutes (give or take a few seconds), it sends out an apparent re-invite, but this time directly to the device’s destination:
[May 9 11:22:20] INVITE sip:TLS1000000@;transport=TLS;rinstance=f61c0ae7d5f16be9 SIP/2.0
This confuses the soft phone and it attempts to create a new session causing audio static until eventually the soft phone sends back an “ACK not received message” – probably because it’s state machine is messed up due to the invite being sent to the wrong place.
If I set “timers=no”, the issue goes away, but I am concerned about running a session without timers.
I do have direct_media=no set as well, so one would assume it would not attempt to even send a re-invite at all.

Any suggestions would be appreciated as to what other configuration parameters I can try – or perhaps this is a bug. This same issue also appears in 13.7.0.

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What is your complete configuration and the full SIP signaling?